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webrtc/audio
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Alex Narest dc9ca9329b Revert "BWE allocation strategy" 
This reverts commit a5fbc23379.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> BWE allocation strategy
> 
> This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test
> 
> Bug: webrtc:8243
> Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac
> Reviewed-on: https://webrtc-review.googlesource.com/13124
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20345}

TBR=stefan@webrtc.org,alexnarest@webrtc.org

Change-Id: I8ed12cd2115ef63204e384cc93c9f4473daa54d1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/14020
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20361}
2017-10-19 15:34:52 +00:00
..
test Fix flag name in low_bandwidth_audio_test.py 2017-10-04 17:26:14 +00:00
utility Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_receive_stream.cc New method RtpReceiver::GetLatestTimestamps. 2017-10-03 16:14:29 +00:00
audio_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_receive_stream_unittest.cc Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 2017-10-02 10:47:00 +00:00
audio_send_stream.cc Revert "BWE allocation strategy" 2017-10-19 15:34:52 +00:00
audio_send_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_send_stream_tests.cc Remove voe_auto_test and add new tests to cover the missing cases. 2017-09-15 16:56:08 +00:00
audio_send_stream_unittest.cc Remove voe::Statistics. 2017-09-29 13:00:28 +00:00
audio_state.cc Remove the VoiceEngineObserver callback interface. 2017-09-26 16:35:01 +00:00
audio_state.h Remove the VoiceEngineObserver callback interface. 2017-09-26 16:35:01 +00:00
audio_state_unittest.cc Remove the VoiceEngineObserver callback interface. 2017-09-26 16:35:01 +00:00
audio_transport_proxy.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_transport_proxy.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
BUILD.gn Fix Gn untracked headers in webrtc/call. 2017-10-10 15:13:48 +00:00
conversion.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
scoped_voe_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00